Hi All,
I am struggling to get H323 to work between Asterisk and Avaya S8400. I want to use Avaya as a tandem switch to reach the PSTN.
I can call from my Avaya digital phone to an xlite sip phone, it rings but when I answer, the xlite shows connected but the call is dropped on the Avaya. I can’t call from the xlite to the Avaya.
The following are copies of my configuration. Any advice?? Thank you.
sip.conf
[general]
context=intern
disallow=all
allow=alaw
allow=ulaw
srvlookup=yes
canreinvite=no
[8390]
type=friend
secret=12345
qualify=yes
host=dynamic
callerid="John Doe"
pickupgroup=1
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
[local]
include=>default
[default]
include=> intern
[intern]
exten => 500,1,voicemailmain(${CALLERIDNUM}@other)
exten => 8390,1,Dial(SIP/8390,20)
exten => 8390,n,voicemail(8390@other,u)
exten => 8390,n,hangup
exten => _8383,1,Dial(OOH323/${EXTEN}@definity)
ooh323.conf
[general]
faststart=yes
h245tunneling=yes
gatekeeper = DISABLE
[definity]
type=friend
context=intern
ip=10.60.10.4
port=1720
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
change signaling-group 84
Page 1
TRUNK GROUP
Group Number: 84 Group Type: isdn CDR Reports: n
Group Name: Asterisk COR: 1 TN: 1 TAC: 805
Direction: two-way Outgoing Display? y Carrier Medium: H.323
Dial Access? n Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 84
Number of Members: 2
Page 2
Group Type: isdn
TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs: 6
Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc
Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:
Disconnect Supervision - In? y Out? y
Answer Supervision Timeout: 0
Page 3
TRUNK FEATURES
ACA Assignment? n Measured: none
Internal Alert? n Maintenance Tests? y
Data Restriction? n NCA-TSC Trunk Member:
Send Name: n Send Calling Number: y
Used for DCS? n Send EMU Visitor CPN? n
Suppress # Outpulsing? n Format: public
UUI IE Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Connected Number: y
Hold/Unhold Notifications? n
Send UUI IE? y Modify Tandem Calling Number? n
Send UCID? n
Send Codeset 6/7 LAI IE? y
Page 4
QSIG TRUNK GROUP OPTIONS
SBS? n
QSIG Value-Added? n
Page 5
TRUNK GROUP
Administered Members (min/max): 1/2
GROUP MEMBER ASSIGNMENTS Total Administered Members: 2
Port Name Night
1: T00011 Asterisk
2: T00012 Asterisk
3:
4:
5:
6:
7:
8:
9:
10:
11:
12:
13:
14:
15:
Change signaling-group 84
SIGNALING GROUP
Group Number: 84 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
IP Video? n Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 84
TSC Supplementary Service Protocol: a
T303 Timer(sec): 10
Near-end Node Name: clana03 Far-end Node Name: asterisk
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 2
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Bypass If IP Threshold Exceeded? n
H.235 Annex H Required? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? y
Link Loss Delay Timer(sec): 90 IP Audio Hairpinning? n
Enable Layer 3 Test? n Interworking Message: PROGress
DCP/Analog Bearer Capability: 3.1kHz
debug 8390 (Asterisk) to 8383 (Avaya)
== Using SIP RTP CoS mark 5
– Executing [8383@intern:1] Dial(“SIP/8390-09d4c9f0”, “OOH323/8383@definity”) in new stack
— ooh323_request - data 8383@definity format 0x8 (alaw)
— find_peer
+++ find_peer
+++ ooh323_request
— ooh323_call- 8383@definity
— onNewCallCreated ooh323c_o_3
— find_call
+++ find_call
setting callid number 8390
Outgoing call definity(ooh323c_o_3) - Codec prefs - (alaw|ulaw)
Adding capabilities to call(outgoing, ooh323c_o_3)
Adding g711 alaw capability to call(outgoing, ooh323c_o_3)
Adding g711 ulaw capability to call(outgoing, ooh323c_o_3)
— configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_3
+++ ooh323_call
– Called 8383@definity
— onCallCleared ooh323c_o_3
— find_call
+++ find_call
— ooh323_hangup
hanging definity
+++ ooh323_hangup
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/8390-09d4c9f0’ status is ‘CHANUNAVAIL’
— ooh323_destroy
Destroying definity
+++ ooh323_destroy
From 8383 to 8390
— onNewCallCreated ooh323c_4
+++ onNewCallCreated ooh323c_4
— ooh323_onReceivedSetup ooh323c_4
— find_user
+++ find_user
sso*CLI>Adding capabilities to call(incoming, ooh323c_4)
— configure_local_rtp
+++ configure_local_rtp
+++ ooh323_onReceivedSetup - Determined context default, extension 8390
— onAlerting ooh323c_4
— find_call
+++ find_call
+++ onAlerting ooh323c_4
– Executing [8390@default:1] Dial(“OOH323/10.60.10.4-308d”, “SIP/8390,20”) in new stack
== Using SIP RTP CoS mark 5
– Called 8390
– SIP/8390-09d51e18 is ringing
----- ooh323_indicate 3 on call ooh323c_4
++++ ooh323_indicate 3 on ooh323c_4
– SIP/8390-09d51e18 answered OOH323/10.60.10.4-308d
— ooh323_answer
+++ ooh323_answer
— onCallEstablished ooh323c_4
— find_call
+++ find_call
+++ onCallEstablished ooh323c_4
----- ooh323_indicate 20 on call ooh323c_4
[Jun 11 19:27:32] WARNING[5544]: chan_ooh323.c:981 ooh323_indicate: Don’t know how to indicate condition 20 on ooh323c_4
++++ ooh323_indicate 20 on ooh323c_4
– Packet2Packet bridging OOH323/10.60.10.4-308d and SIP/8390-09d51e18
----- ooh323_indicate 16 on call ooh323c_4
– Started music on hold, class ‘default’, on OOH323/10.60.10.4-308d
++++ ooh323_indicate 16 on ooh323c_4
== Spawn extension (default, 8390, 1) exited non-zero on ‘OOH323/10.60.10.4-308d’
– Stopped music on hold on OOH323/10.60.10.4-308d
— ooh323_hangup
hanging 10.60.10.4
+++ ooh323_hangup
— onCallCleared ooh323c_4
— find_call
+++ find_call
+++ onCallCleared
— ooh323_destroy
Destroying 10.60.10.4
+++ ooh323_destroy